Can anyone recommend a particular online WebRTC phone for testing with Asterisk?We tried:- JsSIP, but even with the enable video checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with Rejecting secure video stream with..
Author : David Cunningham
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Is there a way to tell in the dialplan, Asterisk AGI, or Asterisk AMI, when a call has been hung up because the SIP rtptimeout has been reached?..
We have a Kamailio and Asterisk cluster, both machines being on a real103.x IP address and also on a 172.x OpenVPN address.The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on its 103.x address, Aster..